The present invention relates generally to the field of communication systems, and more particularly, to text based compression schemes.
Currently, telephony service is provided for the most part over circuit switched networks. A fast emerging new trend called IP telephony provides telephony service over Internet Protocol (IP) networks. The motivating factors for carrying voice traffic over data networks are the integration of voice and data applications, which can result in more effective business process, cost savings for voice calls and enabling of many new services for business and customers. The flexibility offered by IP telephony lies in moving the intelligence from the network to the end stations, thereby enabling many new services that did not exist before. In an effort to merge Internet and cellular telephony, two aspects are focused onxe2x80x94end-to-end call set up delay and voice quality.
Protocols such as Session Initiated Protocol (SIP) and Session Description Protocol (SDP) will typically be used to set up and tear down calls. However, adopting ASCII based protocols such as SIP and SDP in access networks of limited bandwidth incurs a significant delay for call set up; Passing large text messages over the air interface also results in a very inefficient use of the transmission medium. In addition, some legacy based enhanced time division multiplexed (TDM) cellular transceivers, such as GSM EDGE Radio Access Network (GERAN), will need to xe2x80x9cstealxe2x80x9d audio bandwidth in order to transmit in-call SIP signaling messages. This stealing of audio bandwidth will likely result in long audio mutes.
Thus there is a need for a method of compressing text based messages in order to increase spectrum efficiency, reduce transmission delay and provide a comparable level of quality of service compared with circuit switched systems.